Webrtc Phone legacy sip not working

Hi, the Webrtc phone does not work properly with legacy sip, when I created an extent as webrtc using legacy sip the extent can not be registered in asterisk using the webrtc phone because the plugin is trying to register as pjsip. with pjsip work great but in my scenario, I must use legacy SIP

WebSocket connection from 'xx.xx.xx.xx:34672' for protocol 'sip' accepted using version '13'
[2022-02-21 23:20:03] NOTICE[18319]: res_pjsip/pjsip_distributor.c:666 log_failed_request: Request 'REGISTER' from '"204" <sip:204@pbx.com>' failed for 'xx.xx.xx.xx:34672' (callid: aabfeeg7m6b1slvgb3c1o0) - No matching endpoint found
[2022-02-21 23:20:03] NOTICE[18319]: res_pjsip/pjsip_distributor.c:666 log_failed_request: Request 'REGISTER' from '"204" <sip:204@pbx.com>' failed for 'xx.xx.xx.xx:34672' (callid: aabfeeg7m6b1slvgb3c1o0) - No matching endpoint found
[2022-02-21 23:20:03] NOTICE[18319]: res_pjsip/pjsip_distributor.c:666 log_failed_request: Request 'REGISTER' from '"204" <sip:204@pbx.com>' failed for 'xx.xx.xx.xx:34672' (callid: aabfeeg7m6b1slvgb3c1o0) - Failed to authenticate

Any help on how to resolve it will be appreciated.

Comments

  • Webrtc phones knows nothing about channel drivers. Your problem is in your Asterisk configuration.

    There is a setting in sip.conf to enable or disable websocket support. If you want sip websocket to go to chan_sip you will have to set that to true:

    websocket_enabled = true
    Best regards,

  • Thank you for your help that information put me in the right direction, I found the solution.
    This is my working environment
    Issabel4 PBX
    centos 7
    As described in my first post when I created a webrtc extent as chan_sip, and try to use that extent in with the webrtc phone, the PBX does not register the extent, even after I enable websocket_enabled option in the sip advance setting, we need also to enable these options
    tlsenable=yes
    tlsbindaddr=0.0.0.0
    tlscertfile=/etc/asterisk/keys/asterisk.pem
    tlscafile=/etc/asterisk/keys/ca.crt
    tlscipher=ALL
    tlsclientmethod=tlsv1
    those options are not present on the GUI we can add to sip_general_custom.conf
    again thank for your Nicolas

  • Glad it worked.
    Latest issabel version comes with all available webrtc settings for modification (but it is still marked as beta)
    Best regards,

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